U.S. patent application number 12/071005 was filed with the patent office on 2008-07-31 for method and apparatus for enhanced internet telephony.
This patent application is currently assigned to Vonage Holdings Corp.. Invention is credited to Jeffrey Citron, Louis Holder.
Application Number | 20080181375 12/071005 |
Document ID | / |
Family ID | 34465459 |
Filed Date | 2008-07-31 |
United States Patent
Application |
20080181375 |
Kind Code |
A1 |
Holder; Louis ; et
al. |
July 31, 2008 |
Method and apparatus for enhanced internet telephony
Abstract
A method and apparatus for enhanced Internet telephony ensures
that communication between a source and destination is not
interrupted by common network address translation. According to one
aspect of the invention, communication may continue through a
router that employs network address translation.
Inventors: |
Holder; Louis; (Princeton,
NJ) ; Citron; Jeffrey; (Edison, NJ) |
Correspondence
Address: |
John Baird
Suite 1000, 505 9th Street, N.W.
Washington
DC
20004
US
|
Assignee: |
Vonage Holdings Corp.
|
Family ID: |
34465459 |
Appl. No.: |
12/071005 |
Filed: |
February 14, 2008 |
Related U.S. Patent Documents
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Application
Number |
Filing Date |
Patent Number |
|
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10684593 |
Oct 15, 2003 |
|
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12071005 |
|
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Current U.S.
Class: |
379/93.01 |
Current CPC
Class: |
H04L 65/1026 20130101;
H04L 29/12462 20130101; H04L 65/1069 20130101; H04L 65/1006
20130101; H04L 61/2564 20130101; H04L 29/125 20130101; H04L 61/255
20130101 |
Class at
Publication: |
379/93.01 |
International
Class: |
H04M 11/00 20060101
H04M011/00 |
Claims
1-7. (canceled)
8. An internet telephony system configured to use Session
Initiation Protocol (SIP) signaling to setup a communication of
streaming packets, the internet telephony system comprising: a
relay configured to relay streaming packets of the communication
between a caller and a call destination; a server configured to
receive, process and transmit SIP signaling messages to setup the
communication between the caller and the call destination and to
select the relay to use for the communication, the selection being
based at least on the quality of the communication.
9. The internet telephony system of claim 8, wherein the server is
configured to select the relay based in part on the geographic
location of the caller.
10. The internet telephony system of claim 8, wherein the server is
configured to select the relay based in part on the geographic
location of the call destination.
11. The internet telephony system of claim 8, wherein the server is
configured to select the relay to decrease the latency of the
communication.
12. The internet telephony system of claim 8, wherein the server is
configured to select the relay to decrease the travel time of the
communication.
13. The internet telephony system of claim 8, wherein the server is
configured to select the relay to limit the geographical area
traveled by the streaming packets of the communication.
14. The internet telephony system of claim 8, wherein the relay is
associated with a point-of-presence geographically separated from
other points-of-presence.
15. The internet telephony system of claim 8, wherein the server is
configured to select the relay based on a SIP Invite message.
16. The internet telephony system of claim 8, wherein said server
is a pre-proxy server.
17. The internet telephony system of claim 8, wherein said
streaming packet protocol is the Real Time Transport Protocol
(RTP).
18. The internet telephony system of claim 8, wherein the relay is
a RTP relay.
19. The internet telephony system of claim 8, wherein the server is
separated from the relay.
20. A method of providing internet service, the method comprising:
providing a server configured to receive, process and transmit
Session Initiation Protocol (SIP) signaling messages; receiving a
signaling message originating from a caller requesting a
communication to a call destination; selecting a relay for use
during the communication based at least on the quality of the
communication; relaying streaming packets of the communication via
the selected relay between the caller and the call destination.
21. The method of claim 20, wherein the selecting the relay is
based in part on the geographic location of the caller.
22. The method of claim 20, wherein the selecting the relay is
based in part on the geographic location of the call
destination.
23. The method of claim 20, wherein the selecting the relay is
based in part on improving the latency of the communication.
24. The method of claim 20, wherein the selecting the relay is
based in part on improving the travel time of the
communication.
25. The method of claim 20, wherein the selecting the relay is
based in part on limiting the geographical area traveled by the
steaming packets of the communication.
26. The method of claim 20, wherein the relay is associated with a
point-of-presence geographically separated from other
points-of-presence.
27. The method of claim 20, wherein the SIP signaling message
originating from the caller is a SIP Invite.
28. The method of claim 20, wherein the server operates as a
pre-proxy server.
29. The method of claim 20, wherein the streaming packets are Real
Time Transport Protocol (RTP) packets.
30. The method of claim 20, wherein the selected relay is a RTP
relay.
31. The method of claim 20, wherein the server is separated from
the relay.
32. An internet telephony server for setting up a communication of
streaming packets, the server configured to receive and process a
Session Initiation Protocol (SIP) signaling message originated from
a caller requesting a communication of streaming packets to a call
destination, wherein the server is configured to process
information in the SIP signaling message to select a relay from a
plurality of relays available to relay streaming packets of the
communication between the caller and the call destination,
wherein-the server makes the selection based at least on the
quality of the communication.
33. The server of claim 32, further configured to select the relay
based in part on the geographic location of the caller.
34. The server of claim 32, further configured to select the relay
based in part on the geographic location of the call
destination.
35. The server of claim 32, further configured to select the relay
to decrease the latency of the communication.
36. The server of claim 32, further configured to select the relay
to decrease the travel time of the communication.
37. The server of claim 32, further configured to select the relay
to limit the geographical area traveled by the steaming packets of
the communication.
38. The server of claim 32, wherein each relay is associated with a
point-of-presence geographically separated from other
points-of-presence.
39. The server of claim 32, wherein the signaling message is a SIP
Invite.
40. The server of claim 32, wherein the server operates as a
pre-proxy server.
41. The server of claim 32, wherein the streaming packets are Real
Time Transport Protocol (RTP) packets.
42. The server of claim 32, wherein said relay comprises a RTP
relay.
43. The server of claim 32, wherein the server is separated from
the plurality of relays.
Description
[0001] The present application is a divisional application that
claims priority benefit of U.S. patent application Ser. No.
10/684,593 entitled "Method and Apparatus for Enhanced Internet
Telephony", the disclosure of which is hereby incorporated by
reference.
BACKGROUND OF THE INVENTION
[0002] Today, most common residential broadband deployments are
delivered via either cable or DSL modem. Such broadband deployment
typically provides customers with a single Ethernet port that
grants one public IP address to a single computer device. Given
this situation, customers are restricted to using only one
computer, and must purchase a router if they desire to share the
broadband connection to access, for example, the Internet between
more than one of the customer's computer devices.
[0003] To transport media and telephone signaling, customers
commonly use a Multi Media Terminal Adapter (MTA) coupled between
their source device (e.g., a computer or telephone) and their
broadband connection. One example of a common Media Terminal
Adapter is the Cisco ATA 186 Analog to Telephone Adapter (ATA)
manufactured by Cisco Systems, Inc. of San Jose Calif. In the case
of Internet telephony, the Media Terminal Adapter operates as a
handset to Ethernet adapter that converts traditional telephone
analog signals into Internet packets. The packets are then sent
using, for example, a standard protocol such as Session Initiation
Protocol (SIP) on route towards their destination.
[0004] FIG. 1 is a schematic representation of an example user
Internet Telephony environment. In FIG. 1, a cable modem 10
provides access to the Internet 20. In the FIG. 1 example, the user
employs an intermediate communication point, e.g., router 30 to
provide multiple devices access to the Internet 20. The router 30
assigns respective private dynamic IP addresses to the Media
Terminal Adapter 40 and to the computer 50. As shown in the FIG. 1
example, the Media Terminal Adapter 40 is coupled to a common
telephone handset 60. The Media Terminal Adapter 40 receives
signals from the handset 60, creates packets and sends data packets
to the Router 30, which in turn sends them to the cable modem 10
and eventually to the Internet 20.
[0005] A major drawback of the above typical environment is the
difficulty in accommodating the Network Address Translation (NAT)
that is typically implemented by the router 30. As is commonly
understood, a Dynamic Host Configuration Protocol server running on
the router 30 assigns private dynamic IP addresses to the Media
Terminal Adapter 40 and computer 50; thus effecting Network Address
Translation (NAT).
[0006] When a user wishes to initiate a call and activates the
telephone handset 60, the handset sends signals to the Media
Terminal Adapter 40. The Media Terminal Adapter 40 then begins the
communication/registration process with an Internet telephone
service provider. The communication between the Media Terminal
Adapter 40 and a server of the Internet telephone service provider
employs a standard protocol such as Session Initiation Protocol.
But, the router 30 performs the Network Address Translation on a
timed basis. As is commonly known, typical routers used in home
environments assign private IP addresses to devices connected to
the router. But, those addresses are valid only for a limited time.
Thus, after the limited time expires, the private address is no
longer assigned to a given device, such as the Media Terminal
Adapter 40. As a result, the SIP messages sent from Internet
telephone service provider's server are not passed by the router 30
to the Media Terminal Adapter 40. Consequently, the Media Terminal
Adapter 40 can send SIP messages, but is not able to receive
packets from the Internet Telephone service provider's server due
to the router 30 losing the originating outbound port and making
communication to an MTA located behind a router impossible.
[0007] FIG. 2 is a schematic representation of an example
environment that addresses the issue of router 30 losing an
outbound port during an Internet telephone connection. In the FIG.
2 example, at the Internet telephone service provider, a
destination, e.g., a pre-proxy server 70, receives messages from
the router 30. Pre-proxy server 70 records the private IP address
of the Media Terminal Adapter 40 during, for example, the SIP
registration process. It also records the network address
translation communication port assigned by the router 30 to the
Media Terminal Adapter 40 to and from which it will send and
receive messages, such as SIP messages. Upon registration, the
Media Terminal Adapter 40 passes fields used to communicate with
the pre-proxy server 70. Examples of fields, that can be passed
include, for example, the private IP address of the Media Terminal
Adapter 40, the public IP address of the router 30, and port
information. After the pre-proxy server 70 receives the information
from the Media Terminal Adapter 40, the pre-proxy server 70
periodically sends, for example, blank UDP messages to the Media
Terminal Adapter 40, which contain the same destination and source
address as a typical SIP message would have. Other messages could
be used instead of the UDP message. The message used should prompt
the Media Terminal Adapter 40 to send a response to the pre-proxy
server 70. The pre-proxy server 70 sends, for example, the UDP
message to the router 30 using the public IP address of the router
30 and the port information received in the message from the router
30. The pre-proxy server 70 sends, for example, the UDP within the
limited time that the router 30 maintains that private address
assigned to the Media Terminal Adapter. The router 30 accordingly
routes the message to the destination designated in the message
from the pre-proxy server 70. The pre-proxy server 70 also
maintains the private and public IP addresses of the MTA and
rewrites the headers in the actual SIP messages based on this
information.
[0008] The above solution worked, but it did not solve the network
address translation problem for all routers. For example, some
routers would close the outbound port if the device behind the
router's network address translation did not send an outbound
message. Thus, there is a need for a solution to the problem, in
Internet telephony, of the network address translation that a
router performs as a part of its intended operation.
SUMMARY OF THE INVENTION
[0009] It is an object of the present invention to provide a method
and apparatus for enhanced Internet telephony that avoids the above
drawbacks.
[0010] It is another object of the present invention to provide a
method and apparatus for enhanced Internet telephony that allows
the use of Session Initiated Protocol technology.
[0011] It is a further object of the present invention to provide a
method and apparatus for enhanced Internet telephony that allows
the use of Session Initiated Protocol technology within
environments employing network address translation.
[0012] It is still another object of the present invention to
provide a method and apparatus for enhanced Internet telephony that
allows the use of Session Initiated Protocol technology with
routers employing network address translation.
[0013] To achieve the above and other object, the present invention
provides a method for providing enhanced Internet telephony that
includes receiving a message from a source at an intermediate
point; sending at least a portion of the message from the
intermediate point to a destination over the Internet; sending a
response to the message from the destination to the intermediate
point over the Internet; sending the response from the intermediate
point to the source; repeatedly sending other messages from the
destination over the Internet to the intermediate point; sending at
least a portion of corresponding ones of the other messages from
the intermediate point to the source; and sending responses to the
portions of the other messages from the source to the intermediate
point.
BRIEF DESCRIPTION OF THE DRAWINGS
[0014] FIG. 1 is a schematic representation of an example user
Internet Telephony environment.
[0015] FIG. 2 is a schematic representation of an example
environment intended to address the issue of a router losing an
outbound port during an Internet telephone connection.
[0016] FIG. 3 is a schematic representation of an example
environment employing the present invention.
DESCRIPTION OF THE PREFERRED EMBODIMENTS
[0017] FIG. 3 is a schematic representation of an example
environment employing an embodiment of the present invention. In
FIG. 3, a user initiates a call using a telephone handset 60. As
described above, the media terminal adapter 40 implements standard
signaling between itself and an Internet telephony regional data
center 80. Once the user has been registered and the destination
has provided a SIP acknowledgment of the SIP invite sent by the
media terminal adapter 40, communication between the caller 60 and
a customer in a destination area 100 proceeds using, for example,
Real-time Transport Protocol (RTP) between the caller and a
customer in the destination area 100 via the Internet 20 and, for
example, a RTP relay 90 in Internet telephony point-of-presence 110
in the destination area 100.
[0018] However, with the call set up as described above, the router
30 may close the outbound port after a timeout period. As a result,
voice data from the customer in the destination area 110 will not
reach the telephone handset 60 behind router 30. To avoid the
router 30 timing out and closing the outbound port, an embodiment
of the present invention causes the media terminal adapter 40 to
send an outbound message to the Internet telephony regional data
center 80. One way of accomplishing this is to have the pre-proxy
server 75 periodically send an empty SIP notify message to the
media terminal adapter 40. The media terminal adapter 40 responds
to this notify message in accordance with SIP standards by, for
example sending an acknowledgment message. The sending of a message
by the media terminal adapter 40 causes the router 30 to keep the
outbound port open by, for example restarting the router's timeout
period.
[0019] Referring to the exemplary embodiment shown in FIG. 3, the
Internet telephony regional data center 80 has the pre-proxy server
75 separated from the RTP relay 85. While this separation is not
necessary to the present invention, in some environments, it allows
additional functionality to be more easily added to the pre-proxy
server 75. An example of such additional functionality is the
dynamic allocation of the RTP relay 85. The pre-proxy server 75 can
allocate the closest RTP relay between the two calling parties.
That allocation enables the ability to decrease latency and travel
time of the RTP stream. Also as shown in FIG. 3, with the exemplary
embodiment, only SIP messages get routed to the Internet telephony
regional data center 80. The RTP stream need not travel to the data
center, and depending upon the location of the caller and the
destination area 100, can travel within a limited geographic area.
For example, the telephone handset could be located in California,
and the Internet telephony regional data center 80 could be located
in New Jersey. If the destination area 100 is also in California,
the Internet telephony point of presence in the destination area 90
would be allocated by the pre-proxy server 75 to also be in
California. Thus, as noted above, the RTP stream would remain in
California; tending to reduce to latency and travel time of the RTP
stream.
[0020] In the above, the pre-proxy server 75 is shown and discussed
as a separate computer. This is for convenience of discussion, for
purposes of practicing the invention, it does not need to be
separate. Instead, the discussed functions that typically would be
implemented in a pre-proxy server can be implemented in a computer
that is also functioning as a server.
* * * * *